1. Field of the Invention
The present invention relates to a method and an apparatus for transmitting speech signals, and more particularly, to a method and an apparatus for wideband encoding of speech signals and transmission of the encoded bit stream.
2. Description of the Related Art
Conventionally, various coding methods have been proposed to digitize and process speech signals. General speech signal processing methods are classified into two types: in one, 16 bit linear pulse code modulation data is obtained by sampling input analog speech signals at 8 kHz and input to an encoder; in the other, 16 bit linear pulse code modulation data is obtained by sampling input analog speech signals at 16 kHz and input to an encoder. In the former method, speech signals are coded by methods including G.711-G.712 pulse code modulation (PCM) and G.720-G.729 non-PCM standards of the International Telecommunication Union—Telecommunication Standardization Sector (ITU-T). In the latter method, speech signals are coded by G.722 and G.722.1 of the ITU-T and an adaptive multi-rate wideband (AMR-WB) method, which will be used in IMT-2000.
Here, G.723.1, which is a standard for compressing multimedia signals at a lower rate, is an algorithm for compressing and restoring input speech at a dual rate of 5.3/6.3 kbit/s and provides toll quality on a cable network. G.723.1 uses a Hybrid coding technique combining a waveform coding method and a parametric coding method, and is a Code Excited Linear Prediction (CELP)-based speech coder. G.729, which is a standard for IMT-2000 to expand the frequency efficiency for mobile communications, is an algorithm for compressing and restoring input speech at a rate of 8 kbit/s. G.729A is reduced complexity version of the G.729 coder. This version is bitstream interoperable with the full version.
G.729A also provides toll quality on a cable network and uses a Hybrid coding technique and a CELP-based speech coder. ITU-T G.722, which is a standard for coding wideband audio signals, has a transmission rate of 64, 56, or 48 kbit/s and face-to-face communication quality. Also, G.722 divides one band into two sub-bands and codes each of the two sub-bands, using an Adaptive Differential Pulse Code Modulation (AD-PCM) method.
Methods and apparatuses for providing toll quality on a cable network and coding speech at a lower rate have enabled new services in mobile communications and telephony due to high frequency efficiency. In particular, the services using Voice over Internet Protocol (VoIP) over Internet networks are rapidly becoming widespread because of their low telephone rates. However, conventional coding methods and apparatuses have the problem of low service quality due to low toll quality and long delay over Internet networks. Thus, the conventional coding methods and apparatuses do not have a good reputation.
Accordingly, in order to solve these problems, various attempts have been made. For example, if speech signals are sampled at a frequency of 16 kHz in a VoIP system before coding, the quality of the speech signals may be much improved. However, current 16 kHz wideband speech codecs are not at all compatible with the codec currently used in the VoIP service. Thus, a new system is needed. Also, since wideband signals have a wide frequency bandwidth, networks with a large data processing capacity are needed. Thus, starting new services in disregard of current systems has many difficulties.